1. Field of the Invention
The invention relates to a sample rate converter for filtering a discrete-time input signal having a sample rate q.multidot.f.sub.s by means of a system filter having an impulse response h(t) and, besides, for converting the discrete-time input signal having a sample rate q.multidot.f.sub.s to a discrete-time output signal having a sample rate p.multidot.f.sub.s, p and q being unequal positive integers greater than or equal to one, the sample rate converter including a cascade combination of a partial filter and an equalizer for equalizing the frequency characteristic of the sample rate converter.
2. Description of the Related Art
A sample rate converter of this type is known from the journal article entitled "Area-Efficient Multichannel Oversampled PCM Voiceband Coder" in IEEE Journal of Solid State Circuits, Vol. 23, No. 6, December 1988.
Sample rate converters of this type are used, for example, in analog-to-digital converters and digital-to-analog converters operating according to the sigma-delta principle. In a sigma-delta analog-to-digital converter the analog input signal is converted to a digital 1-bit signal that denotes the sign of the difference between a sample of the analog input signal and a sample of suitably filtered preceding 1-bit signals.
The sample rate of the sigma-delta analog-to-digital converter is many times higher than the minimum required sample rate according to Shannon's sampling theorem. The advantage of this is that the anti-aliasing filter which is to reduce the bandwidth of the analog input signal to half the sample rate may now be arranged in a much simpler manner in that the passband and the stopband of this anti-aliasing filter are much wider apart than in the case where the sample rate were about equal to the minimum sample rate required according to the sampling theorem.
However, in many cases a digital output signal is ultimately desired to have a sample rate which is about equal to the minimum sample rate required according to the sampling theorem. This lower sample rate is often desired for further processing the, for example, digital output signal with the aid of a bit-parallel arranged signal processor which, in addition, has a limited processing rate. Also for the transmission of such a signal by, for example, a telephone line, the sample rate of the digital signal is desired not to be higher than is strictly necessary.
For obtaining a reduction of the sample rate, the 1-bit signal is applied to a sample rate converter which derives a reduced sample rate PCM signal from the 1-bit signal.
A known property (known, for example, from aforementioned article) of sigma-delta modulators is that the 1-bit signal comprises quantizing noise with a frequency-dependent spectral power density, which spectral power density strongly increases with frequency. If the conversion of the 1-bit signal to a PCM signal having a reduced sample rate is effected, for example, by adding bits over a specific time interval, the high-frequency quantizing noise in the 1-bit signal will be aliased to the baseband in which the desired signal is situated. Consequently, the signal-to-noise ratio of the PCM signal is degraded considerably.
In order to avoid this degradation of the signal-to-noise ratio, the noise of the 1-bit signal, which noise has frequencies exceeding the maximum frequency of the desired PCM signal, is to be eliminated with the aid of a filter to be termed rest filter hereinafter, before the sample rate is reduced.
A similar problem occurs when the sample rate is increased from a first sample rate f.sub.1 to a second sample rate f.sub.2. Once the sample rate has been increased, the frequency spectrum of the discrete-time signal having the second sample rate continues to be periodic with a period f.sub.1 that corresponds to the first sample rate, whereas a signal is desired which has a frequency spectrum that is only periodic with a period f.sub.2 that corresponds to the second (higher) sample rate. In order to realise this, a system filter is required which also eliminates the undesired frequency components between 1/2 f.sub.1 and 1/2 f.sub.2.
For a reduction of the complexity of the system filter, a system filter having an impulse response h(t) which is simple to realize is chosen for the known sample rate converter. As a result, the frequency characteristic of the system filter in the passband is not flat, so that the frequency spectrum of the baseband signal will change. In order to realize a flat frequency characteristic in the passband of the whole sample rate converter, the sample rate converter is arranged as a cascade configuration of a partial filter and an equalizer, while the whole system filter or part thereof is incorporated in the partial filter.
Although the use of a partial filter and an equalizer leads to some reduction of the complexity of the sample rate converter, the need for reducing this complexity still continues to exist.